In the area of human-machine speech interface, or in hands-free telecommunication such as audio phones, it is usually desired to process only the voice of the speaker(s) that are close to a microphone, and ignore background noise. Some degree of interference rejection can be achieved through the use of a voice detector, such as the ones described in U.S. patent application Ser. No. 09/375,309, entitled "METHOD FOR ENHANCEMENT OF ACOUSTIC SIGNAL IN NOISE" and U.S. patent application Ser. No. 09/385,975, entitled "SYSTEM AND METHOD FOR CLASSIFICATION OF SOUND SOURCES", both of which are assigned to the assignee of the present invention. However, such voice detectors still let voice interference's, such as remote conversations, television sets, and public announcement systems, be processed.
Most prior art approaches rely on sound volume (loudness) to determine whether a sound source is sufficiently near the microphone to warrant processing it. However, even though the volume of a source is somewhat correlated to its distance to a microphone, a distant loud source can often be perceived as louder than a weaker, albeit closer source.
Another way to determine the range of an acoustic source is to use triangulation through the use of several pairs of microphones. This approach is computationally onerous, and necessitates much significant additional hardware.
The inventor has determined that it would be desirable to be able to estimate the range of a sound source independently of its inherent loudness using only two microphones. The present invention provides a system and method for determining the range of an acoustic signal within a reverberant space that avoids the limitations of prior techniques.